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niklase@google.com470e71d2011-07-07 08:21:25 +00001/*
pwestin@webrtc.orgf6bb77a2012-01-24 17:16:59 +00002 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
niklase@google.com470e71d2011-07-07 08:21:25 +00003 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020011#ifndef MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
12#define MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
niklase@google.com470e71d2011-07-07 08:21:25 +000013
Yves Gerey988cc082018-10-23 12:03:01 +020014#include <stddef.h>
15#include <stdint.h>
Jonas Olssona4d87372019-07-05 19:08:33 +020016
nisse14adba72017-03-20 03:52:39 -070017#include <memory>
danilchapb8b6fbb2015-12-10 05:05:27 -080018#include <set>
Steve Anton296a0ce2018-03-22 15:17:27 -070019#include <string>
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000020#include <vector>
niklase@google.com470e71d2011-07-07 08:21:25 +000021
Danil Chapovalovd264df52018-06-14 12:59:38 +020022#include "absl/types/optional.h"
Yves Gerey988cc082018-10-23 12:03:01 +020023#include "api/rtp_headers.h"
Erik Språngeeaa8f92018-05-17 12:35:56 +020024#include "api/video/video_bitrate_allocation.h"
Yves Gerey988cc082018-10-23 12:03:01 +020025#include "modules/include/module_fec_types.h"
26#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020027#include "modules/rtp_rtcp/include/rtp_rtcp.h"
Yves Gerey988cc082018-10-23 12:03:01 +020028#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" // RTCPPacketType
Tomas Gunnarsson593e6a42020-06-07 22:32:31 +020029#include "modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.h"
Erik Språngbfcfe032021-08-04 14:45:32 +020030#include "modules/rtp_rtcp/source/packet_sequencer.h"
Yves Gerey988cc082018-10-23 12:03:01 +020031#include "modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020032#include "modules/rtp_rtcp/source/rtcp_receiver.h"
33#include "modules/rtp_rtcp/source/rtcp_sender.h"
Erik Språng77b75292019-10-28 15:51:36 +010034#include "modules/rtp_rtcp/source/rtp_packet_history.h"
Erik Språng9c771c22019-06-17 16:31:53 +020035#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020036#include "modules/rtp_rtcp/source/rtp_sender.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020037#include "rtc_base/gtest_prod_util.h"
Markus Handellf7303e62020-07-09 01:34:42 +020038#include "rtc_base/synchronization/mutex.h"
niklase@google.com470e71d2011-07-07 08:21:25 +000039
niklase@google.com470e71d2011-07-07 08:21:25 +000040namespace webrtc {
41
Yves Gerey988cc082018-10-23 12:03:01 +020042class Clock;
43struct PacedPacketInfo;
44struct RTPVideoHeader;
45
Tommi3a5742c2020-05-20 09:32:51 +020046// DEPRECATED.
danilchap59cb2bd2016-08-29 11:08:47 -070047class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
pwestin@webrtc.org2853dde2012-05-11 11:08:54 +000048 public:
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +020049 explicit ModuleRtpRtcpImpl(
50 const RtpRtcpInterface::Configuration& configuration);
Danil Chapovalov2a5ce2b2018-02-07 09:38:31 +010051 ~ModuleRtpRtcpImpl() override;
niklase@google.com470e71d2011-07-07 08:21:25 +000052
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000053 // Returns the number of milliseconds until the module want a worker thread to
54 // call Process.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000055 int64_t TimeUntilNextProcess() override;
niklase@google.com470e71d2011-07-07 08:21:25 +000056
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000057 // Process any pending tasks such as timeouts.
pbosa26ac922016-02-25 04:50:01 -080058 void Process() override;
niklase@google.com470e71d2011-07-07 08:21:25 +000059
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000060 // Receiver part.
niklase@google.com470e71d2011-07-07 08:21:25 +000061
stefan@webrtc.orga5cb98c2013-05-29 12:12:51 +000062 // Called when we receive an RTCP packet.
nisse479d3d72017-09-13 07:53:37 -070063 void IncomingRtcpPacket(const uint8_t* incoming_packet,
64 size_t incoming_packet_length) override;
pwestin@webrtc.org1da1ce02011-10-13 15:19:55 +000065
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000066 void SetRemoteSSRC(uint32_t ssrc) override;
Tommi08be9ba2021-06-15 23:01:57 +020067 void SetLocalSsrc(uint32_t ssrc) override;
wu@webrtc.org822fbd82013-08-15 23:38:54 +000068
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000069 // Sender part.
Niels Möller5fe95102019-03-04 16:49:25 +010070 void RegisterSendPayloadFrequency(int payload_type,
71 int payload_frequency) override;
Peter Boström8b79b072016-02-26 16:31:37 +010072
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000073 int32_t DeRegisterSendPayload(int8_t payload_type) override;
asapersson@webrtc.org5249cc82011-12-16 14:31:37 +000074
Johannes Kron9190b822018-10-29 11:22:05 +010075 void SetExtmapAllowMixed(bool extmap_allow_mixed) override;
76
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000077 // Register RTP header extension.
Sebastian Janssonf39c8152019-10-14 17:32:21 +020078 void RegisterRtpHeaderExtension(absl::string_view uri, int id) override;
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000079 int32_t DeregisterSendRtpHeaderExtension(RTPExtensionType type) override;
Sebastian Janssonf39c8152019-10-14 17:32:21 +020080 void DeregisterSendRtpHeaderExtension(absl::string_view uri) override;
niklase@google.com470e71d2011-07-07 08:21:25 +000081
Mirko Bonadei999a72a2019-07-12 17:33:46 +000082 bool SupportsPadding() const override;
83 bool SupportsRtxPayloadPadding() const override;
stefan53b6cc32017-02-03 08:13:57 -080084
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000085 // Get start timestamp.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000086 uint32_t StartTimestamp() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +000087
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000088 // Configure start timestamp, default is a random number.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000089 void SetStartTimestamp(uint32_t timestamp) override;
niklase@google.com470e71d2011-07-07 08:21:25 +000090
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000091 uint16_t SequenceNumber() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +000092
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000093 // Set SequenceNumber, default is a random number.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000094 void SetSequenceNumber(uint16_t seq) override;
niklase@google.com470e71d2011-07-07 08:21:25 +000095
Per83d09102016-04-15 14:59:13 +020096 void SetRtpState(const RtpState& rtp_state) override;
97 void SetRtxState(const RtpState& rtp_state) override;
98 RtpState GetRtpState() const override;
99 RtpState GetRtxState() const override;
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +0000100
Ivo Creusen8c40d512021-07-13 12:53:22 +0000101 void SetNonSenderRttMeasurement(bool enabled) override {}
102
Erik Språng6841d252019-10-15 14:29:11 +0200103 uint32_t SSRC() const override { return rtcp_sender_.SSRC(); }
niklase@google.com470e71d2011-07-07 08:21:25 +0000104
Amit Hilbuch77938e62018-12-21 09:23:38 -0800105 void SetRid(const std::string& rid) override;
106
Steve Anton296a0ce2018-03-22 15:17:27 -0700107 void SetMid(const std::string& mid) override;
108
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000109 void SetCsrcs(const std::vector<uint32_t>& csrcs) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000110
henrik.lundin@webrtc.org1972ff82014-09-11 06:20:28 +0000111 RTCPSender::FeedbackState GetFeedbackState();
niklase@google.com470e71d2011-07-07 08:21:25 +0000112
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000113 void SetRtxSendStatus(int mode) override;
114 int RtxSendStatus() const override;
Erik Språngc06aef22019-10-17 13:02:27 +0200115 absl::optional<uint32_t> RtxSsrc() const override;
mflodman@webrtc.org9f5ebb52013-04-12 14:55:46 +0000116
Shao Changbine62202f2015-04-21 20:24:50 +0800117 void SetRtxSendPayloadType(int payload_type,
118 int associated_payload_type) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000119
Danil Chapovalovd264df52018-06-14 12:59:38 +0200120 absl::optional<uint32_t> FlexfecSsrc() const override;
brandtr9dfff292016-11-14 05:14:50 -0800121
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000122 // Sends kRtcpByeCode when going from true to false.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000123 int32_t SetSendingStatus(bool sending) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000124
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000125 bool Sending() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000126
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000127 // Drops or relays media packets.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000128 void SetSendingMediaStatus(bool sending) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000129
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000130 bool SendingMedia() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000131
Erik Språng1e51a382019-12-11 16:47:09 +0100132 bool IsAudioConfigured() const override;
133
Sebastian Jansson1bca65b2018-10-10 09:58:08 +0200134 void SetAsPartOfAllocation(bool part_of_allocation) override;
135
Niels Möller5fe95102019-03-04 16:49:25 +0100136 bool OnSendingRtpFrame(uint32_t timestamp,
137 int64_t capture_time_ms,
138 int payload_type,
139 bool force_sender_report) override;
140
Erik Språng9c771c22019-06-17 16:31:53 +0200141 bool TrySendPacket(RtpPacketToSend* packet,
142 const PacedPacketInfo& pacing_info) override;
143
Erik Språng1d50cb62020-07-02 17:41:32 +0200144 void SetFecProtectionParams(const FecProtectionParams& delta_params,
145 const FecProtectionParams& key_params) override;
146
147 std::vector<std::unique_ptr<RtpPacketToSend>> FetchFecPackets() override;
148
Erik Språnga9229042019-10-24 12:39:32 +0200149 void OnPacketsAcknowledged(
150 rtc::ArrayView<const uint16_t> sequence_numbers) override;
stefan@webrtc.org0a3c1472013-12-05 14:05:07 +0000151
Erik Språngf6468d22019-07-05 16:53:43 +0200152 std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
153 size_t target_size_bytes) override;
Erik Språng478cb462019-06-26 15:49:27 +0200154
Erik Språng3663f942020-02-07 10:05:15 +0100155 std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
156 rtc::ArrayView<const uint16_t> sequence_numbers) const override;
157
Erik Språng04e1bab2020-05-07 18:18:32 +0200158 size_t ExpectedPerPacketOverhead() const override;
159
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000160 // RTCP part.
niklase@google.com470e71d2011-07-07 08:21:25 +0000161
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000162 // Get RTCP status.
pbosda903ea2015-10-02 02:36:56 -0700163 RtcpMode RTCP() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000164
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000165 // Configure RTCP status i.e on/off.
pbosda903ea2015-10-02 02:36:56 -0700166 void SetRTCPStatus(RtcpMode method) override;
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000167
168 // Set RTCP CName.
Peter Boström9ba52f82015-06-01 14:12:28 +0200169 int32_t SetCNAME(const char* c_name) override;
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000170
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000171 // Get remote NTP.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000172 int32_t RemoteNTP(uint32_t* received_ntp_secs,
173 uint32_t* received_ntp_frac,
174 uint32_t* rtcp_arrival_time_secs,
175 uint32_t* rtcp_arrival_time_frac,
176 uint32_t* rtcp_timestamp) const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000177
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000178 // Get RoundTripTime.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000179 int32_t RTT(uint32_t remote_ssrc,
180 int64_t* rtt,
181 int64_t* avg_rtt,
182 int64_t* min_rtt,
183 int64_t* max_rtt) const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000184
Niels Möller5fe95102019-03-04 16:49:25 +0100185 int64_t ExpectedRetransmissionTimeMs() const override;
186
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000187 // Force a send of an RTCP packet.
188 // Normal SR and RR are triggered via the process function.
Erik Språng242e22b2015-05-11 10:17:43 +0200189 int32_t SendRTCP(RTCPPacketType rtcpPacketType) override;
190
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000191 void GetSendStreamDataCounters(
asapersson@webrtc.org97d04892014-12-09 09:47:53 +0000192 StreamDataCounters* rtp_counters,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000193 StreamDataCounters* rtx_counters) const override;
asapersson@webrtc.org97d04892014-12-09 09:47:53 +0000194
Henrik Boström6e436d12019-05-27 12:19:33 +0200195 // A snapshot of the most recent Report Block with additional data of
196 // interest to statistics. Used to implement RTCRemoteInboundRtpStreamStats.
197 // Within this list, the ReportBlockData::RTCPReportBlock::source_ssrc(),
198 // which is the SSRC of the corresponding outbound RTP stream, is unique.
199 std::vector<ReportBlockData> GetLatestReportBlockData() const override;
Alessio Bazzicabc1c93d2021-03-12 17:45:26 +0100200 absl::optional<SenderReportStats> GetSenderReportStats() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000201
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000202 // (REMB) Receiver Estimated Max Bitrate.
Danil Chapovalov1de4b622017-12-13 13:35:10 +0100203 void SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) override;
Danil Chapovalov51e21aa2017-10-10 17:46:26 +0200204 void UnsetRemb() override;
pwestin@webrtc.org741da942011-09-20 13:52:04 +0000205
danilchap59cb2bd2016-08-29 11:08:47 -0700206 void SetTmmbn(std::vector<rtcp::TmmbItem> bounding_set) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000207
nisse284542b2017-01-10 08:58:32 -0800208 size_t MaxRtpPacketSize() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000209
nisse284542b2017-01-10 08:58:32 -0800210 void SetMaxRtpPacketSize(size_t max_packet_size) override;
michaelt79e05882016-11-08 02:50:09 -0800211
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000212 // (NACK) Negative acknowledgment part.
niklase@google.com470e71d2011-07-07 08:21:25 +0000213
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000214 // Send a Negative acknowledgment packet.
philipel83f831a2016-03-12 03:30:23 -0800215 // TODO(philipel): Deprecate SendNACK and use SendNack instead.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000216 int32_t SendNACK(const uint16_t* nack_list, uint16_t size) override;
stefan@webrtc.org6a4bef42011-12-22 12:52:41 +0000217
philipel83f831a2016-03-12 03:30:23 -0800218 void SendNack(const std::vector<uint16_t>& sequence_numbers) override;
219
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000220 // Store the sent packets, needed to answer to a negative acknowledgment
221 // requests.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000222 void SetStorePacketsStatus(bool enable, uint16_t number_to_store) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000223
Per Kjellander16999812019-10-10 12:57:28 +0200224 void SendCombinedRtcpPacket(
225 std::vector<std::unique_ptr<rtcp::RtcpPacket>> rtcp_packets) override;
226
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000227 // Video part.
Elad Alon7d6a4c02019-02-25 13:00:51 +0100228 int32_t SendLossNotification(uint16_t last_decoded_seq_num,
229 uint16_t last_received_seq_num,
Elad Alone86af2c2019-06-03 14:37:50 +0200230 bool decodability_flag,
231 bool buffering_allowed) override;
Elad Alon7d6a4c02019-02-25 13:00:51 +0100232
Erik Språngbf46cfe2020-05-11 18:22:02 +0200233 RtpSendRates GetSendRates() const override;
234
danilchap59cb2bd2016-08-29 11:08:47 -0700235 void OnReceivedNack(
236 const std::vector<uint16_t>& nack_sequence_numbers) override;
237 void OnReceivedRtcpReportBlocks(
238 const ReportBlockList& report_blocks) override;
239 void OnRequestSendReport() override;
henrike@webrtc.orgf5da4da2012-02-15 23:54:59 +0000240
Erik Språng566124a2018-04-23 12:32:22 +0200241 void SetVideoBitrateAllocation(
242 const VideoBitrateAllocation& bitrate) override;
sprang5e38c962016-12-01 05:18:09 -0800243
Niels Möller5fe95102019-03-04 16:49:25 +0100244 RTPSender* RtpSender() override;
245 const RTPSender* RtpSender() const override;
246
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000247 protected:
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000248 bool UpdateRTCPReceiveInformationTimers();
niklase@google.com470e71d2011-07-07 08:21:25 +0000249
Erik Språng77b75292019-10-28 15:51:36 +0100250 RTPSender* rtp_sender() {
Erik Språng9cdc9cc2019-10-28 18:24:32 +0100251 return rtp_sender_ ? &rtp_sender_->packet_generator : nullptr;
Erik Språng77b75292019-10-28 15:51:36 +0100252 }
253 const RTPSender* rtp_sender() const {
Erik Språng9cdc9cc2019-10-28 18:24:32 +0100254 return rtp_sender_ ? &rtp_sender_->packet_generator : nullptr;
Erik Språng77b75292019-10-28 15:51:36 +0100255 }
nissea33c62e2017-03-14 00:49:45 -0700256
257 RTCPSender* rtcp_sender() { return &rtcp_sender_; }
258 const RTCPSender* rtcp_sender() const { return &rtcp_sender_; }
259
260 RTCPReceiver* rtcp_receiver() { return &rtcp_receiver_; }
261 const RTCPReceiver* rtcp_receiver() const { return &rtcp_receiver_; }
262
Tomas Gunnarsson79ca92d2020-06-18 17:30:15 +0200263 void SetMediaHasBeenSent(bool media_has_been_sent) {
264 rtp_sender_->packet_sender.SetMediaHasBeenSent(media_has_been_sent);
265 }
266
Sebastian Janssonaa01f272019-01-30 11:28:59 +0100267 Clock* clock() const { return clock_; }
nissea33c62e2017-03-14 00:49:45 -0700268
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000269 private:
asapersson@webrtc.orge7b1e112013-12-16 14:40:36 +0000270 FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, Rtt);
asapersson@webrtc.org1ae1d0c2013-11-20 12:46:11 +0000271 FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, RttForReceiverOnly);
niklase@google.com470e71d2011-07-07 08:21:25 +0000272
Erik Språng77b75292019-10-28 15:51:36 +0100273 struct RtpSenderContext {
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +0200274 explicit RtpSenderContext(const RtpRtcpInterface::Configuration& config);
Erik Språng77b75292019-10-28 15:51:36 +0100275 // Storage of packets, for retransmissions and padding, if applicable.
Erik Språng9cdc9cc2019-10-28 18:24:32 +0100276 RtpPacketHistory packet_history;
Erik Språngbfcfe032021-08-04 14:45:32 +0200277 // Handles sequence number assignment and padding timestamp generation.
278 PacketSequencer sequencer_;
Erik Språng77b75292019-10-28 15:51:36 +0100279 // Handles final time timestamping/stats/etc and handover to Transport.
Tomas Gunnarsson593e6a42020-06-07 22:32:31 +0200280 DEPRECATED_RtpSenderEgress packet_sender;
Erik Språng77b75292019-10-28 15:51:36 +0100281 // If no paced sender configured, this class will be used to pass packets
Artem Titov913cfa72021-07-28 23:57:33 +0200282 // from `packet_generator_` to `packet_sender_`.
Tomas Gunnarsson593e6a42020-06-07 22:32:31 +0200283 DEPRECATED_RtpSenderEgress::NonPacedPacketSender non_paced_sender;
Erik Språng77b75292019-10-28 15:51:36 +0100284 // Handles creation of RTP packets to be sent.
Erik Språng9cdc9cc2019-10-28 18:24:32 +0100285 RTPSender packet_generator;
Erik Språng77b75292019-10-28 15:51:36 +0100286 };
287
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000288 void set_rtt_ms(int64_t rtt_ms);
289 int64_t rtt_ms() const;
asapersson@webrtc.org1ae1d0c2013-11-20 12:46:11 +0000290
asapersson@webrtc.orgba8138b2014-12-08 13:29:02 +0000291 bool TimeToSendFullNackList(int64_t now) const;
292
Niels Mölleraf6ea0c2020-11-20 12:21:21 +0100293 // Returns true if the module is configured to store packets.
294 bool StorePackets() const;
295
296 // Returns current Receiver Reference Time Report (RTTR) status.
297 bool RtcpXrRrtrStatus() const;
298
Erik Språng77b75292019-10-28 15:51:36 +0100299 std::unique_ptr<RtpSenderContext> rtp_sender_;
300
nisse150708e2017-03-16 05:02:53 -0700301 RTCPSender rtcp_sender_;
302 RTCPReceiver rtcp_receiver_;
303
Sebastian Janssonaa01f272019-01-30 11:28:59 +0100304 Clock* const clock_;
nisse150708e2017-03-16 05:02:53 -0700305
asapersson@webrtc.org9ffd8fe2015-01-21 08:22:50 +0000306 int64_t last_bitrate_process_time_;
307 int64_t last_rtt_process_time_;
sprang168794c2017-07-06 04:38:06 -0700308 int64_t next_process_time_;
asapersson@webrtc.org9ffd8fe2015-01-21 08:22:50 +0000309 uint16_t packet_overhead_;
niklase@google.com470e71d2011-07-07 08:21:25 +0000310
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000311 // Send side
Danil Chapovalov9eb6ce12017-12-15 12:25:01 +0100312 int64_t nack_last_time_sent_full_ms_;
asapersson@webrtc.orgba8138b2014-12-08 13:29:02 +0000313 uint16_t nack_last_seq_number_sent_;
stefan@webrtc.org9354cc92012-06-07 08:10:14 +0000314
Niels Möller5fe95102019-03-04 16:49:25 +0100315 RemoteBitrateEstimator* const remote_bitrate_;
316
Tommi5f223652018-03-26 13:28:26 +0200317 RtcpRttStats* const rtt_stats_;
asapersson@webrtc.org1ae1d0c2013-11-20 12:46:11 +0000318
319 // The processed RTT from RtcpRttStats.
Markus Handellf7303e62020-07-09 01:34:42 +0200320 mutable Mutex mutex_rtt_;
Niels Möllercd982132020-11-26 16:19:56 +0100321 int64_t rtt_ms_ RTC_GUARDED_BY(mutex_rtt_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000322};
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000323
324} // namespace webrtc
325
Mirko Bonadei92ea95e2017-09-15 06:47:31 +0200326#endif // MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_